and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines
Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)
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r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
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r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
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r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines
Merged revisions 119237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines
- Instead of only enforcing destination call number checking on an ACK, check
all full frames except for PING and LAGRQ, which may be sent by older versions
too quickly to contain the destination call number.
(As suggested by Tim Panton on the asterisk-dev list)
- Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ
from being sent before the destination call number is known.
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A lot of whitespace issues have been resolved in this commit
Also some doc updates, but that's only 6 lines
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
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r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines
Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP.
(closes issue #12501)
Reported by: slimey
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r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines
Realtime flag affects construction in multiple ways, so consulting whether
rtcachefriends was set was done too soon (needed to be done inside build_peer,
not just as a flag to build_peer).
Also, fullcontact needed to be reconstructed, because realtime separates the
embedded ';' into multiple fields.
(closes issue #12722)
Reported by: barthpbx
Patches:
20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
Tested by: barthpbx
(Much of the discussion happened on #asterisk-dev for diagnosing this issue)
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them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647)
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) | 3 lines
Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure.
Another commit is following to make sure the zt_chan_conf structure is not modified.
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'unknown', and better document the use of each parameter.
(closes issue #12633)
Reported by: tzafrir
Patches:
pridialplan_unknown_2.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
random window.
This commit is essentially a workaround for some undesirable behaviour of SDL;
we should not be doing this in the application, but in the library.
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supplied window.
SDL (at least recent 1.2.x versions) has the ability to run the
graphic output into an externally supplied window, whose ID in the
environment variable SDL_WINDOWID. Ideally, applications should
run unchanged irrespective of who creates the window. Unfortunately,
SDL does not subscribe to mouse, key and resize events on externally
supplied windows, so we need to do ask for these events explicitly.
On passing, also add some code to handle SDL_ACTIVEEVENT so if
the X11 window is killed while we are active, we call
"stop now" to terminate the asterisk instance.
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r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines
Avoid access of uninitialized memory. This caused a bunch of crashes for me
while doing load testing of development branch where I'm working on some
performance improvements.
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r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines
Fix a deadlock involving channel autoservice and chan_local that was debugged
and fixed by mmichelson and me.
We observed a system that had a bunch of threads stuck in ast_autoservice_stop().
The reason these threads were waiting around is because this function waits to
ensure that the channel list in the autoservice thread gets rebuilt before the
stop() function returns. However, the autoservice thread was also locked, so
the autoservice channel list was never getting rebuilt.
The autoservice thread was stuck waiting for the channel lock on a local channel.
However, the local channel was locked by a thread that was stuck in the autoservice
stop function.
It turned out that the issue came down to the local_queue_frame() function in
chan_local. This function assumed that one of the channels passed in as an
argument was locked when called. However, that was not always the case. There
were multiple cases in which this channel was not locked when the function was
called. We fixed up chan_local to indicate to this function whether this channel
was locked or not. The previous assumption had caused local_queue_frame() to
improperly return with the channel locked, where it would then never get unlocked.
(closes issue #12584)
(related to issue #12603)
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marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines
Merged revisions 115564 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines
Fix a race condition that bbryant just found while doing some IAX2 testing.
He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes,
however, the audio was extremely choppy. We looked at a packet trace and saw
a storm of INVAL and VNAK frames being sent from one box to another.
It turned out that what had happened was that one box tried to send a CONTROL
frame before the 3 way handshake had completed. So, that frame did not include
the destination call number, because it didn't have it yet. Part of our recent
work for security issues included an additional check to ensure that frames that
are supposed to include the destination call number have the correct one. This
caused the frame to be rejected with an INVAL. The frame would get retransmitted
for forever, rejected every time ...
This race condition exists in all versions that got the security changes,
in theory. However, it is really only likely that this would cause a problem in
Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_
beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing
all versions that could potentially be affected by the introduced race condition.
These changes are what bbryant and I came up with to fix the issue. Instead of
simply dropping control frames that get sent before the handshake is complete,
the code attempts to wait a little while, since in most cases, the handshake
will complete very quickly. If it doesn't complete after yielding for a little
while, then the frame gets dropped.
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r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines
Avoid putting opaque="" in Digest authentication. This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)
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dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.
(closes issue #12536)
Reported by: bjm
Patches:
12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm
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r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
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r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined
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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines
Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.
(closes issue #12513)
Reported by: mneuhauser
Patches:
asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)
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(closes issue #12496)
Reported by: daniele
Patches:
misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
Tested by: daniele
Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision.
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r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line
added fix from #8899
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r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines
If the dial string passed to the call channel callback does not indicate an
extension, then consider the extension on the channel before falling back
to the default.
(closes issue #12479)
Reported by: darren1713
Patches:
exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116)
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
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r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines
Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon
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r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines
use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10)
(closes issue #12456)
Reported by: fnordian
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r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines
The call_token on the pvt can occasionally be NULL, causing a crash.
If it is NULL, we can skip this channel, since it can't the one we're looking for.
(closes issue #9299)
Reported by: vazir
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r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines
Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen.
Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed.
(issue #12400)
Reported by: ztel
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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines
We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.
(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann
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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines
If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor
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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines
If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.
(closes issue #12392)
Reported by: fnordian
Patches:
chan_sip.patch uploaded by fnordian (license 110) with small modification from me
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines
Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes.
I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short.
Instead of freeing the struct and nulling the pointer, it now just resets it, because this
ast_str is expected by the calling routine to still be there after handle_request_do() returns.
This appears to fix the crash. I assume that it was introduced with ast_str's being adopted. It's a subtle and easy-to-miss sort of problem.
I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well;
no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines
Fix the testing of the "res" variable so that it is more logically correct and
makes the correct warning and debug messages print.
(closes issue #12361)
Reported by: one47
Patches:
chan_zap_deferred_digit.patch uploaded by one47 (license 23)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
that valgrind no longer complains and that calls do complete correctly.
The fix is along the same lines as before: Make sure the final null terminator gets copied
into the new sip_request's data pointer. Without it, parse_request will read and potentially
write past the end of the string, causing potential crashes.
(closes issue #12284...for real this time!)
reported by falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.
(closes issue #12284)
Reported by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.
2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.
3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.
4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.
5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.
(closes issue #11968)
Reported by: dimas
Patches:
v2-11968-dsp.patch uploaded by dimas (license 88)
v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines
When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.
So this is a revert of a revert...sort of.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.
(closes issue #12279)
Reported by: rjain
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines
This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk
so that all scheduler functions are fixed at once.
I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.
(closes issue #12272)
Reported by: qq12345
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines
Merged revisions 110335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines
Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address
structure that a background thread continuously updates. However, in these cases,
a stack variable was passed. That means that the dnsmgr thread would be continuously
writing to bogus memory.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
by Juggie while attempting to load chan_zap. Apparently this would happen
if an error were encountered while trying to process zapata.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when dialing a channel that does not provide progress (analog ZAP lines)
The phone does handle the double update on calls to channels that do
provide progress and wont insert duplicate items
(closes issue #12239)
Reported by: DEA
Patches:
chan_skinny-call-log.txt uploaded by DEA (license 3)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Brett Bryant
Russell Bryant
Joshua Colp
Philippe Sultan
Christian Richter
Michiel van Baak
Jeff Peeler
Sean Bright
Luigi Rizzo
Matthew Fredrickson
Jim Dixon
Olle Johansson
Mark Michelson
Jason Parker
Kevin P. Fleming
Steve Murphy
Terry Wilson